| 05:31.17 | *** join/#asterisk infobot (ibot@rikers.org) |
| 05:31.17 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.0 (2016/11/23), 11.25.0 (2016/11/23), Standard: 14.2.0 (2016/11/23); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 05:41.05 | *** join/#asterisk infobot (ibot@rikers.org) |
| 05:41.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.0 (2016/11/23), 11.25.0 (2016/11/23), Standard: 14.2.0 (2016/11/23); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 05:53.56 | *** join/#asterisk infobot (ibot@rikers.org) |
| 05:53.56 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.0 (2016/11/23), 11.25.0 (2016/11/23), Standard: 14.2.0 (2016/11/23); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:26.00 | bittis | goodmorning everyone |
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| 09:09.02 | GeekNerd_phone | farts |
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| 13:21.41 | kannan | hello, i would like to emulate a sed find & replace in asterisk logger. I would like to be able to perform the replace operation prior to being written in syslog. instead of hard coding, perhaps to add a filter entry in logger.conf where the pattern regex and replacement string can be defined . where should i start ? |
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| 13:31.15 | kannan | hello, i would like to emulate a sed find & replace in asterisk logger. I would like to be able to perform the replace operation prior to being written in syslog. instead of hard coding, perhaps to add a filter entry in logger.conf where the pattern regex and replacement string can be defined . where should i start ? |
| 13:36.28 | foobarbarian_ | Hi, it's been driving me crazy. The operator switched to Huawai x3000 and prob. ever since I can't do incoming calls anymore. The call is working fine until asterisk terminated it with "Retransmission timeout reached on transmission..." There's no firewall inbetween |
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| 13:51.35 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:52.34 | [TK]D-Fender | foobarbarian_, Show us the call |
| 13:52.38 | [TK]D-Fender | ~pb |
| 13:52.38 | infobot | rumour has it, pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:52.39 | [TK]D-Fender | ^^^ |
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| 14:13.47 | foobarbarian_ | [TK]D-Fender: tcpdump or asterisk -rvvv ? |
| 14:16.04 | WIMPy | Asterisk with sip debug on. |
| 14:16.08 | kannan | logger provides an exec_after_rotation, but this still has a time period when undesirable resides in syslog, ditto if i run sed or a perl job to do the same. how is it possible to transform text prior to writing to syslog. |
| 14:17.04 | WIMPy | Edit the source, |
| 14:17.39 | kannan | WIMPy , thanks, could you point me in the right place to look? |
| 14:18.21 | WIMPy | makes a wild guess at logger.c. |
| 14:18.26 | kannan | ideally, i would like to add a filter with pattern & replacement options via logger.conf |
| 14:18.29 | kannan | ok thanks |
| 14:19.16 | foobarbarian_ | http://pastebin.com/jZhVDKsr this is mobile->asterisk->lync I can hear the lync server on the mobile for around 5 seconds until asterisk breaks up the connection because it thinks the connection is broken. I've already tried pedantic=yes (the only suggestion i could find on google) |
| 14:20.14 | foobarbarian_ | The asterisk server is directly connected to the trunk, no firewall or nat in between |
| 14:24.18 | foobarbarian_ | I'm getting this error even tho the connection is already established and I can hear the recorded message: [2016-12-08 22:14:39] WARNING[2137]: chan_sip.c:4066 retrans_pkt: Hanging up call sfidl32eljjckcfnljgpmco2i3mokegj@SoftX3000 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 14:26.58 | foobarbarian_ | That's the trunk config: http://pastebin.com/3Gv9zjpy |
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| 14:42.58 | [TK]D-Fender | --- SIP read from UDP:10.202.142.36:5060 ---> |
| 14:42.58 | [TK]D-Fender | INVITE sip:027796912@snapcomparemanila.pldt.net:5060;user=phone SIP/2.0 |
| 14:43.03 | [TK]D-Fender | User-Agent: Huawei SoftX3000 V300R010 |
| 14:43.15 | [TK]D-Fender | You should not have a local device listed as being NAT'd like thatt |
| 14:43.27 | [TK]D-Fender | <--- Transmitting (NAT) to 10.202.142.36:5060 ---> |
| 14:43.27 | [TK]D-Fender | SIP/2.0 100 Trying |
| 14:43.30 | [TK]D-Fender | As we see in the response |
| 14:43.38 | [TK]D-Fender | Fix your peer |
| 14:47.57 | foobarbarian_ | What should I change? nat=no? |
| 14:49.36 | foobarbarian_ | [TK]D-Fender: the call is being established. I can hear around 5-10 seconds of recording until asterisk hangs up on me |
| 14:49.51 | [TK]D-Fender | Do it |
| 14:50.16 | foobarbarian_ | [TK]D-Fender: Already had nat=no but will try again |
| 14:50.37 | [TK]D-Fender | clearly you didn't based on that response I pasted |
| 14:50.48 | [TK]D-Fender | <[TK]D-Fender> <--- Transmitting (NAT) to 10.202.142.36:5060 ---> |
| 14:50.49 | [TK]D-Fender | ^^^^ |
| 14:51.34 | foobarbarian_ | [TK]D-Fender: Sorry I'm being stupid. But nat=no is already set |
| 14:52.01 | [TK]D-Fender | not in the right place |
| 14:52.05 | [TK]D-Fender | or applied |
| 14:52.40 | foobarbarian_ | I'll try and remove the line. I'm not sure what's happening. the trunk worked before the provider chaged |
| 14:54.18 | foobarbarian_ | Same problem. Maybe i'm misunderstanding your instructions? sorry about that |
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| 15:03.18 | foobarbarian_ | [TK]D-Fender: what I'm not understanding is why the call works already until asterisk decidedes it doesn't |
| 15:03.50 | foobarbarian_ | [TK]D-Fender: the nat=no is set for the trunk, is there another place to set it? |
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| 15:41.40 | Gugge | foobarbarian_: remove it from the peer having 10.202.142.36 |
| 15:42.11 | Samot | insecure=very <--- incorrect. |
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| 19:20.37 | BeachBall | new install - my sip phone is constantly ringing with unknow callers |
| 19:20.43 | BeachBall | how to secure my asterisk? |
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| 20:08.35 | mmlj4 | how indeed |
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| 20:32.25 | dan_j_ | Hi. I keep getting this weird error and I'm wondering whether it's someone trying to crash/hack asterisk or if it's somewhere in my config. Although nothings being reloaded when this error appears. |
| 20:32.26 | dan_j_ | http://pastebin.com/T0xTcUH8 |
| 20:33.07 | dan_j_ | Any idea what could be causing that? |
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| 20:58.59 | jfindley | is it normal for a fastagi connection to die when playing back a long audio file? |
| 21:02.26 | igcewieling | jfindley: using php? |
| 21:02.38 | jfindley | python, but same should be the case for any language |
| 21:02.48 | igcewieling | PHP' |
| 21:03.20 | igcewieling | PHP's max execution time defaults to a min or two. |
| 21:06.20 | jfindley | it's just the AGI connection dying |
| 21:06.31 | jfindley | the audio keeps playing, but once it's done further agi calls fail because the socket has been closed |
| 21:07.01 | jfindley | looks like it drops off after 37 seconds |
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| 21:14.54 | rmudgett | dan_j: Looks like you have an IPv4 address and port inside an IPv6 address and port: [31.104.38.247:38852]:38852 |
| 21:28.26 | dan_j | Weird. I dont have any ipv6 addresses listed. I'll check my config for that ip though |
| 21:28.46 | jfindley | igcewieling: I figured it out. It wasn't asterisk. |
| 21:43.16 | *** topic/#asterisk by kharwell -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.1 (2016/12/08), 11.25.1 (2016/12/08), Standard: 14.2.1 (2016/12/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 21:58.34 | Phrohdoh | Am I using WaitTime incorrectly or possibly misunderstanding the point? https://bpaste.net/show/4d473e1c8583 My understanding is that 2 seconds after connection, if someone hasn't answered the call, the call should disconnect. Is this wrong? |
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| 22:28.38 | rmudgett | ~book |
| 22:28.38 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 22:52.38 | [sr] | hi WIMPy |
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| 23:29.32 | wyoung | WIMPy!!!!!!!1 |
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